The area of communication has evolved rapidly over the last years, going from traditional person-to-person phone calls to more advanced services, such as multiparty video conferencing. These services put extensive requirements on the transport network and when media like audio and video are sent over those networks, it is not uncommon that the capacity is lower than what is required to give the end user an ultimate user experience, such as for video telephony over a 3G (third generation) network. Here high definition video of several Mbps (Megabit per second) could be needed to deliver a real high quality experience, while on the other hand, such a high bitrate could only be supported under benign conditions.
Moreover, to compensate for variations in time between received packets, a jitter buffer is used in the receiver. There is a balance in the size of the jitter buffer, which is called the depth of the jitter buffer. The jitter buffer should be deep enough to allow most or all packets to arrive prior to presentation, but if it is too deep, the delay introduced by the jitter buffer reduces the quality of how the real-time communication is perceived.
Also, as a result of the varying network conditions, a video telephony service with a high fixed bitrate over mobile accesses will likely lead to quality problems and unsatisfied users. To mitigate this some services (like Skype and Apple Facetime) have implemented mechanisms to cope with temporarily congested networks, thus providing bitrate adaptation. Through various techniques they try to adapt the media stream bitrate to suit the transport channel.
However, when the bitrate adaptation increases a bitrate, this may cause increased jitter, which may result in the jitter buffer operation being insufficient.